This application is related to the field of digital communications and more specifically to iteratively improving the performance of digital communication systems which use forward error correction and interleaving to overcome the effects of communication channels.
The area of digital communications has undergone a significant transformation in the last ten years because of the discovery of turbo codes. As is known in the art, turbo codes are a class of error correction codes which enable reliable communications with power efficiencies close to the theoretical limit. Turbo codes incorporate two fundamental concepts: concatenated coding and iterative decoding. In order to achieve power efficiencies close to theoretical limits, Soft In Hard Out (SIHO) devices typical to most communication systems, such as Viterbi decoders, were replaced with Soft In Soft Out devices (SISO). These devices allowed for dramatic improvements in performance as soft information was shared and improved on each iteration. However, computational complexity was also increased.
The “turbo” concept can be applied to more than just Forward Error-Correction (FEC) code schemes. For example, waveforms developed for the High Frequency (HF) band conventionally place an interleaver between the FEC code scheme and the transmitted symbols. The interleaver re-sorts the data bits of the FEC encoded data stream to provide a time-diversity in the bit-pattern to separate adjacent bits. Time-diversity is advantageous as it de-correlates errors that are introduced in digital signals by slow fading channels and/or multipath. If the interleaver is not used, the correlation of adjacent errors contributes to rendering the FEC encoding schemes ineffective as adjacent errors may not be corrected.
FIG. 1 illustrates a conventional digital communication system 100 using a “turbo” concept. In this system, transmit signal 110 is applied to encoder 115, which applies a Forward Error Correcting (FEC) code to signal 110. The FEC encoded signal is then applied to interleaver 120 to impose time-diversity into the encoded bit-stream. The encoded, time-diverse signal is then applied to modulator 125, which organizes individual bits into symbols based on the transmission system characteristics. For example, in a QPSK system two (2) bits are selected for each transmission symbol. Similarly, in a 16-QAM system four (4) bits are selected for each transmission symbol. The transmission symbols are then applied to transmission Digital Low-Pass Filter 130, which removes high frequency signal components that may be induced by switching instantaneously from one symbol to the next symbol. The symbols are next applied to Digital Up-Converter 135 to up-convert the symbols to a conventional carrier frequency, which is then band-limited by Transmit Filter 140. In a conventional HF communication system, for example, the up-converted carrier signal is 1800 Hz with a bandwidth of 3 kHz (up-conversion to 1800 Hz creates a real audio signal that can be sent to a radio). The up-converted signal is then transmitted over a wireless communication network or channel represented as communication cloud 145.
Channel 145 may be characterized as a multipath, time-varying environment that produces both time and frequency dispersion of the transmitted signal. For example, one source of multipath in long-haul HF communications is reflections of signals from different layers in the ionosphere. Another example is multiple reflections that occur between the earth's surface and the ionosphere. This gives rise to an effect known as multi-hop propagation of the transmitted signal. Accordingly, the received signal may include several echoes or modes, separated in time by a matter of milliseconds, i.e., time dispersion and/or may experience a frequency dispersion. Frequency dispersion occurs when each received signal is itself fading due to the nature of the ionosphere reflection. In some channels, the multipath or delay spread can range up to six (6) milliseconds and the fading rate or Doppler Spread can be as high as 5 Hz.
The transmitted signal, when received by a receiving system, is applied to Receiver Filter 150, which limits the bandwidth of the received signal to a bandwidth commensurate with the transmission bandwidth. The bandwidth-limited received signal is then applied to Digital Down-Converter 155 and Digital Low-Pass filter 160. The down-converted signal is then demodulated by demodulator 165, de-interleaved by deinterleaver 170 and FEC decoded by decoder 175, in well known processes that remove the modulation, interleaving and encoding performed at the transmitter. Decoded output signal 180 represents an estimate of transmit signal 110 as the communication path induced-errors contribute to the receiving system incorrectly determining the value of a transmitted bit or symbol.
Significant advances have been made in the estimation of a transmitted symbol by using devices that generate soft information at the output of the device in the demodulation process (i.e., SISO devices). However, generating soft output information requires new Soft-in, Soft-out (SISO) devices that produce soft outputs rather than hard decisions, i.e., one or zero (SIHO devices), regarding the transmitted symbol or bit. Examples of SISO devices are the Soft Output Viterbi Algorithm (SOVA) or Maximum A Posteriori (MAP) algorithm, e.g., Bahl Coche Jelinek Raviv (BCJR), which are computationally more complex than conventional hard decision devices, such as the well known Viterbi decoder, Reed-Solomon Decoder, Golay decoder, etc. Hence, the improvement in the estimation of transmitted signals, measured as system performance, has been obtained with increased complexity in processing and cost of soft decision devices. Furthermore, a new equalizer device would be needed to replace a standard equalizer since it would need to be able to use hard or soft information from the decoder to improve its performance (standard equalizers have no way of using this additional information). Note that standard equalizers are SISO devices since their input is the received data and their output symbol estimates.
Thus, there is a need for a system that provides improved system performance while using less costly and less computationally complex devices. In addition to the computational savings, the ability to use well understood (mature and standard) equalizer and decoder algorithms is a significant advantage of this invention.
Accordingly, it is an object of the present invention to provide a novel system and method for determining channel equalization factors to improve, for example, the reception performance of a transmitted signal.
It is another object of the present invention to provide a novel system and method for determining channel equalization factors to improve, for example, the reception performance of a transmitted signal over a digital communication channel.
It is yet another object of the present invention to provide a novel method for determining channel equalization factors to improve, for example, the reception performance of a transmitted signal by determining at least one estimate associated with a decision of a received signal corresponding to the transmitted signal, modulating the estimate, and determining the equalization factors dependent upon the received signal and modulated estimate.
It is still another object of the present invention to provide a novel system for determining channel equalization factors to be applied to a digital channel equalizer to improve, for example, the reception performance of a signal transmitted over a digital communication channel.
It is a further object of the present invention to provide a novel system for determining channel equalization factors to be applied to a digital channel equalizer to improve, for example, the reception performance of a signal transmitted over a digital communication channel where the system includes circuitry that determines at least one estimate associated with a decision of a received signal corresponding to the transmitted signal, circuitry that modulates the estimate, and circuitry that determines the equalization factors dependent upon the received signal and modulated estimate.
It is to be understood that these drawings are solely for purposes of illustrating the concepts of the invention and are not intended as a definition of the limits of the invention. The embodiments shown in FIGS. 1 through 9 and described in the accompanying detailed description are to be used as illustrative embodiments and should not be construed as the only manner of practicing the invention. Also, the same reference numerals, possibly supplemented with reference characters where appropriate, have been used to identify similar elements.